Changes to Incoming Call Processing

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Aaron
Site Admin
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Joined: Thu Jul 12, 2007 12:13 am

Changes to Incoming Call Processing

Post by Aaron » Mon Nov 19, 2007 5:44 am

Hi All,

The "Use dial plan for incoming calls" option has now been made redundant and it has been removed from the interface. The mode of operation is now that the dial plan will be checked for an entry matching your username and if found it will be used. If a matching entry is not found the defualt behaviour is to call all SIP devices registered with the sipswitch for that username.

The "Use dial plan for incoming calls" was causing too much confusion and the new mode of operation defaults to expected behaviour without removing any functionality.

In addition multiple contacts can now be recognised as belonging to a sipswitch user account for incoming calls. The approach for this is to register the contact name as:

<custom string>.<username>@sip.mysipswitch.com

The sipswitch will match on the <username> portion to determine the owner and then lookup an entry in their dialplan for <custom string>.<username>. If no entry is found the default behaviour is again to use the registered devices.

Regards,

Aaron

yf
Posts: 64
Joined: Tue Aug 28, 2007 9:08 am

Urgent Urgent Urgent

Post by yf » Tue Nov 20, 2007 3:53 pm

Aaron:
For sure whatever setting you did caused my entire account not to be able to make outgoing and incoming calls. Something went wrong and I am unable to understand what you are saying nor be able to nail down the problem.

The registered sip accounts for incoming calls are callcentric, GIZMO, voxalot. It used to work very smoothly with no problem until yesterday.

Also, I am unable to make any out calls. I have the following dialplans for outgoing:
exten =~ .*,1,SwitchCall(username,password,${dst}@us.voxalot.com)
exten =~ ^1,1,Switch(username,password,${dst}@callcentric.com)
exten =~ ^011,1,Switch(username,password,${dst}@callcentric.com)
exten =~ ^*18,1,Switch(username,password,${dst}@us.voxalot.com)

They are not working right now.

I also have the following to record calls (which so far did not work).
SwitchCall(username, password,${dst}@us.voxalot.com,,,true)

Can anyone help me please and tell me what is going on?

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Nov 20, 2007 3:57 pm


yf
Posts: 64
Joined: Tue Aug 28, 2007 9:08 am

Post by yf » Tue Nov 20, 2007 4:18 pm

Aaron:
Believe me I spent all day reading all the postings and I did not understand a word.

Where should I insert
<custom string>.<username>@sip.mysipswitch.com

Is this in the outgoing dial plan or the contact for incoming.

What custom strings should be used and when I should dial it?

emoci
Posts: 127
Joined: Mon Aug 20, 2007 11:27 pm

Post by emoci » Tue Nov 20, 2007 6:20 pm

yf wrote:Aaron:
Believe me I spent all day reading all the postings and I did not understand a word.

Where should I insert
<custom string>.<username>@sip.mysipswitch.com

Is this in the outgoing dial plan or the contact for incoming.

What custom strings should be used and when I should dial it?
The only change that's happened affects Incoming Calls only. In the registration section (bottom of config page) you can now specify the contact adress as:
user@sip.mysipswitch.com
or
AnyPrefix.user@sip.mysipswitch.com

If that's all you do, than both those SIP URIs ring the same MySipSwitch Account.

However if you were to define a DIAL PLAN like this:

exten = AnyPrefix.user,1,SwitchCall(user,pass,Number@provider.com)

Now all calls to user@sip.mysipswitch.com (and all DIDs pointed here) will continue to ring your account.

However calls to AnyPrefix.user@sip.mysipswitch.com and any DIDs pointed here will actually ring your Cell/Landline number you specified above.

Outgoing plans and calls are not affected by the change though....

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Wed Nov 21, 2007 4:46 am

Further Modification:

Incoming calls will only use a dial plan entry if there is an exact match on the contact. This means anyone not using the dial plan for incoming calls should not have to concern themselves with the custom contact strings etc. and can use their ATA to receive calls as they were previously.

An example of an exact match for a username of 333999 is:

exten => 333999,1,SwitchCall(..)

An example of an expression match (that would not get used for an incoming call) is:

exten => _3X.,1,SwitchCall(..)

The only exception is if you happen to need a dial plan rule for an outgoing call that exactly matches the contact used for one of your registrations. I suspect this condition is pretty rare if there are any at all.

Regards,

Aaron

yf
Posts: 64
Joined: Tue Aug 28, 2007 9:08 am

Post by yf » Thu Nov 22, 2007 7:15 am

Aaron:
Thank you for your continuous unfailing support. Actually, the reason for my posting was a complete failure for the VoIP system took place on Nov 19 and Nov 20. Something went wrong with all the SIP providers. The calls were behaving in a strange way with all SIP providers. While I can make calls to PSTN numbers I was not able to make calls between SIP numbers. This issue was not related to me only but other friends who were using Mysipswitch. Your email of changing the outgoing dialplan came out on the same day. So I thought at the beginning that it relates to my problem which was not. When checking the websites of all the SIP providers I am using I found no announcement of any kind. I started to track where the problem is by eliminating my ATA and move to softphone, then using each account individually. I realized at the end that this was a major SIP issue across the providers I am using. After two days, things came back to normal. I apologize if my posting revealed some of my frustrations at that time.

I am very interested in the changes made in the outgoing dialplan. It adds more potentials to Mysipswitch which I realized how powerful tool it is during the last two days.

I don’t have the enough technical background. I was introduced to the VoIP and SIP by accident two years ago and since then I am teaching myself. I moved from just a user to commercial VoIP companies to have my own ATA and built up my own SIP network. I still feel that my technical knowledge is very limited and fragile. The postings on the web are for professionals. Even the ATA manuals such as Linksys are written for professionals. I wish we have a 123SIP or SIP for dummies books. So be patient for any dummy question I may ask.

Thanks.

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TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Thu Nov 22, 2007 5:14 pm

If you want, i can help too, i also started with VoIP recently, don't be embarressed, VoIP is a very new technology, i bought some books on the subject, and reading them in my time off moments at work, one book even starts at mr.Bell's invention :) it's fun reading, is my opinion ...you get an insight of the mechanics, your own voice generation, voice codecs, and the transport over the internet....and NAT......:(
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

yf
Posts: 64
Joined: Tue Aug 28, 2007 9:08 am

Post by yf » Thu Nov 22, 2007 11:39 pm

Thank you TheFug
I start learning VoIP in my spare time during the last two years. Right now I don't have enough time. In a couple of month I will start having spare time.

At the moment I will like to learn the programing of dialplans with mysipswitch. It will be useful if users post samples of their dialplans (outgoing and incoming) and explains what it does.

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TheFug
Posts: 914
Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Fri Nov 23, 2007 12:57 am

exten =~ ^(18),1,Switch(fwdusername,fwdpassword,*18${EXTEN:2}@fwd.pulver.com)
exten => _00X.,1,Switch(vbusername,vbpassword,${EXTEN}@sip.voipbuster.com)
exten => _06X.,1,Switch(vcusername,vcpassword,+316${EXTEN:2}@sip.VoipCheap.com)
exten => _*X.,1,Switch(fwdusername,fwdpassword,${EXTEN}@fwd.pulver.com)
exten => _08X.,1,Switch(fwdusername,fwdpassword,*318${EXTEN:2}@fwd.pulver.com)
exten => _0ZX.,1,Switch(vbusername,vbpassword,+31${EXTEN:1}@sip.voipbuster.com)



okay, the first line makes it possible for me to dial all toll-free numbers in the USA, via FreeWorld Dialup, {EXTEN2} does not dial out the first 2 key strokes on phone, but does dial *18 and the 3rd keystrokes and there after keyed in numbers. * is needed by FWD.

The second line detects that i dialed 00 and does not change that and a provider is selected to call long distance cheap rate landline numbers.

the 3rd line sees i'm dialing 06 and dials 00316 and the rest i keyed in after i keyed in: 06 with a provider that does cheap GSM calls for me, al GSM nubers in Nederlands start with 06

The last line detects 0Z where Z can be any number in the range 1-9
these calls are to dutch cities within the Dutch borders, 0031 is the country code for the Netherlands, the following 2 digits, (without the 0) is the cityregio dial code.

Incomming dial plans i have to learn yet....
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

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